The present invention is directed to a method for transmitting voice information between subscriber stations of a radio system. The present invention is also directed to an arrangement for the implementation of the method, as well as, to a subscriber station and a central station of a corresponding transmission system.
Given transmission of voice information between subscriber stations of a radio system, particularly a mobile radio telephone service, it is well known to sample analog voice information at the transmission side and to convert it into digitally coded voice information in a voice encoder and to transmit it, potentially error-protected. At the reception side, the digitally coded voice information is in turn converted into analog voice information and is then output as voice.
In a mobile radio telephone service, an area to be covered is usually divided into a large plurality of cells. Respective base stations are provided in the cells, and the connections between a public telephone network and the mobile radio equipment of the individual subscribers in a respective cell can be set up over these base stations. Such a mobile radio telephone system is, for example, the GSM (global system for mobile communication) standardized by the European Telecommunication Standards Institute (ETSI). Such a system is known from a publication of Siemens AG, D 900 Mobile Communication System, System Description, 1992 or from a publication by M. Boehm, Schaller, W., Mobilfunksystem CD 900, Funk-Technik 411, No. 4, 1986, pp. 150-153. The system identified as DCS 1800/PCN is a similar system.
In the GSM, the voice at the interface between the base stations and the mobile radio equipment is transmitted with digitally coded signals. A coding algorithm is thereby employed for the voice that implements data compression by a factor of 8 to 13 kbit/s.
In order to protect the data stream against disturbances on the radio channel, an error correction method is applied wherein redundancy is attached by channel encoding of the information to be transmitted. As a result of this channel encoding, the voice data rate is then raised from 13 kbit/s to 22.8 kbit/s.
In order, on the one hand, to be able to correct accidental bit errors and, on the other hand, bundling errors of the transmission channel, the transmitted data is also interleaved. They are divided into subblocks and transmitted. At the reception side, the subblocks are collected and deinterleaved. Thereafter, the data are supplied to an error correction means that implements an error correction of the received data and the redundancy attached at the transmission side is removed. The data obtained in this manner is supplied to the voice decoder that reconstructs the linearly quantized samples. A following digital-to-analog converter converts the digital signals into analog voice.
The time required for the transmission of a voice frame of 20 ms from a mobile radio telephone user to a fixed network subscriber, or vice versa, amounts to approximately 90 ms. Given voice transmission from a mobile radio telephone subscriber to another mobile radio telephone subscriber, this delay time is doubled and amounts to approximately 180 ms. European reference EP 04 44 592 A2 also discloses that voice frames of M subscribers are transmitted in a frame having N channels of a mobile radio telephone system. The quality of the transmitted voice, however, can suffer as a result of echoes due to this long delay time.
In order to improve the quality of the voice transmission, it is important to shorten this delay time. It would be conceivable to transmit the channel-coded voice frame in a smaller plurality of sub-blocks, whereby the plurality of bits in the sub-blocks is then increased. In this method, however, too many bits are lost in case of disturbances on the mobile radio channel, i.e. the error-protection method is not adequate given an unaltered data rate.